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/*
* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef AUDIO_AUDIO_RECEIVE_STREAM_H_
#define AUDIO_AUDIO_RECEIVE_STREAM_H_
#include <memory>
#include <vector>
#include "api/audio/audio_mixer.h"
#include "api/neteq/neteq_factory.h"
#include "api/rtp_headers.h"
#include "api/sequence_checker.h"
#include "audio/audio_state.h"
#include "call/audio_receive_stream.h"
#include "call/syncable.h"
#include "modules/rtp_rtcp/source/source_tracker.h"
#include "system_wrappers/include/clock.h"
namespace webrtc {
class PacketRouter;
class ProcessThread;
class RtcEventLog;
class RtpPacketReceived;
class RtpStreamReceiverControllerInterface;
class RtpStreamReceiverInterface;
namespace voe {
class ChannelReceiveInterface;
} // namespace voe
namespace internal {
class AudioSendStream;
class AudioReceiveStream final : public webrtc::AudioReceiveStream,
public AudioMixer::Source,
public Syncable {
public:
AudioReceiveStream(Clock* clock,
RtpStreamReceiverControllerInterface* receiver_controller,
PacketRouter* packet_router,
ProcessThread* module_process_thread,
NetEqFactory* neteq_factory,
const webrtc::AudioReceiveStream::Config& config,
const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
webrtc::RtcEventLog* event_log);
// For unit tests, which need to supply a mock channel receive.
AudioReceiveStream(
Clock* clock,
RtpStreamReceiverControllerInterface* receiver_controller,
PacketRouter* packet_router,
const webrtc::AudioReceiveStream::Config& config,
const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
webrtc::RtcEventLog* event_log,
std::unique_ptr<voe::ChannelReceiveInterface> channel_receive);
AudioReceiveStream() = delete;
AudioReceiveStream(const AudioReceiveStream&) = delete;
AudioReceiveStream& operator=(const AudioReceiveStream&) = delete;
~AudioReceiveStream() override;
// webrtc::AudioReceiveStream implementation.
void Reconfigure(const webrtc::AudioReceiveStream::Config& config) override;
void Start() override;
void Stop() override;
bool IsRunning() const override;
webrtc::AudioReceiveStream::Stats GetStats(
bool get_and_clear_legacy_stats) const override;
void SetSink(AudioSinkInterface* sink) override;
void SetGain(float gain) override;
bool SetBaseMinimumPlayoutDelayMs(int delay_ms) override;
int GetBaseMinimumPlayoutDelayMs() const override;
std::vector<webrtc::RtpSource> GetSources() const override;
// AudioMixer::Source
AudioFrameInfo GetAudioFrameWithInfo(int sample_rate_hz,
AudioFrame* audio_frame) override;
int Ssrc() const override;
int PreferredSampleRate() const override;
// Syncable
uint32_t id() const override;
absl::optional<Syncable::Info> GetInfo() const override;
bool GetPlayoutRtpTimestamp(uint32_t* rtp_timestamp,
int64_t* time_ms) const override;
void SetEstimatedPlayoutNtpTimestampMs(int64_t ntp_timestamp_ms,
int64_t time_ms) override;
bool SetMinimumPlayoutDelay(int delay_ms) override;
void AssociateSendStream(AudioSendStream* send_stream);
void DeliverRtcp(const uint8_t* packet, size_t length);
const webrtc::AudioReceiveStream::Config& config() const;
const AudioSendStream* GetAssociatedSendStreamForTesting() const;
private:
static void ConfigureStream(AudioReceiveStream* stream,
const Config& new_config,
bool first_time);
AudioState* audio_state() const;
SequenceChecker worker_thread_checker_;
webrtc::AudioReceiveStream::Config config_;
rtc::scoped_refptr<webrtc::AudioState> audio_state_;
SourceTracker source_tracker_;
const std::unique_ptr<voe::ChannelReceiveInterface> channel_receive_;
AudioSendStream* associated_send_stream_ = nullptr;
bool playing_ RTC_GUARDED_BY(worker_thread_checker_) = false;
std::unique_ptr<RtpStreamReceiverInterface> rtp_stream_receiver_;
};
} // namespace internal
} // namespace webrtc
#endif // AUDIO_AUDIO_RECEIVE_STREAM_H_
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